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研究生:黃啟榮
研究生(外文):Chi-Jung Huang
論文名稱:SIP網路電話核心技術之研究
論文名稱(外文):The Research on SIP-Based VoIP Core Technology
指導教授:黃紹華黃紹華引用關係
口試委員:林政源馬尚智葉偉和何文楨葉政育
口試日期:2012-05-21
學位類別:博士
校院名稱:國立臺北科技大學
系所名稱:電機工程系博士班
學門:工程學門
學類:電資工程學類
論文種類:學術論文
論文出版年:2012
畢業學年度:100
語文別:英文
論文頁數:67
中文關鍵詞:SIPNAT
外文關鍵詞:SIPNAT
相關次數:
  • 被引用被引用:3
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網路的普及以及電信業者骨幹網路的提升,使得網路愛好者日益增加,加上行動裝置的助長,網路電話被應用的機會也就大大增加,所以我們就此對SIP核心做了些研討及改進。
我們的系統除了解決了SIP協定上的瑕疵,以及Client side和Server side整體的效能提升,使伺服器可以有最大的容量,並且提供使用者可以在嚴苛的環境下繼續能使用網路電話。
SIP通訊協定只能使用UDP-5060及TCP-5060,但在許多網路環境中只提供TCP-80,我們有鑒如此,開發出HTTP方式的SIP通訊協定,不管是對伺服器的訓令或是語音串流都可以使用HTTP方式,克服網路被封鎖的問題。
在開始Media Session時,影像傳輸資料量及其龐大,使用Relay Server來做轉送,不僅需要較大的頻寬及多台Relay Sever來做分流,頻寬及設備的成本都是筆龐大的支出,我們使用Prot Prediction的方式成功達到了Direct Peer to Peer,解決頻寬及設備的費用問題。
SIP通訊協定中使用頻率最高的程序是Registration,這程序的目的除了是身分驗證外還有定位的功能,使得其他使用者可以順利撥打,但是太過於頻繁註冊對於伺服器來說也是一個負擔,太過於忙碌去處理定位而沒有辦法服務撥打,這就本末倒置了,我們的伺服器開發出自動偵測使用者NAT socket關閉時間,用最有效率的方式及不改變SIP制定標準下,準確掌握NAT的Keep-alive time interval,延長使用者註冊時間,讓使用者及伺服器保持連線達到最佳效率。在我們的大量實驗結果,使用者因該要使用TCP封包來與伺服器溝通且Media Session中Audio Streaming或Video Streaming封包尺寸越大越好。
以上的改進使得總體通話達到90%以上成功DP2P,Proxy Server的總體用戶量提升31倍,Media Relay Server最大同時可服務達到3524通電話。本論文的分析結果和改進方法,可以大大提升VoIP系統整體性能及降低成本。


Networks are widely used, and carrier backbone networks have been upgraded. The increase in speed has led to more internet users. Therefore, more opportunities are available to use VoIP products, including IP-phones, video phones, video conferencing, and IP-PBX. This study focused on improving the SIP core.
Our system overcomes the defects of the SIP protocol and enhances the overall performance of the client side and server side. The server allows a large number of users, and enables users to use VoIP in harsh environments.
The SIP protocol provides only UDP-5060 and TCP-5060. However, several network environments support only TCP-80. Therefore, we developed the SIP protocol using the HTTP method. The HTTP method can be used to send messages or stream. This method overcomes the problem of network blockages.
Video streaming data are large in media sessions. The use of relay server forwarding enables the setup of numerous servers; however, it requires considerable bandwidth. The bandwidth and equipment costs are high. We used port prediction to achieve direct peer to peer connections and reduce the bandwidth and equipment costs.
The most frequently used process in the SIP protocol is registration. The purpose of this program is authentication and positioning. It allows users to invite other users. However, a large number of registrations can burden the server. If the server is busy managing the positioning, it cannot service a call session, which is unacceptable. Our server automatically detects the NAT socket closing time of the user. This is the most efficient approach, and does not change the SIP standard-setting or accurate knowledge of the NAT keep-alive time interval. To extend user registrations, users and servers maintain connections for optimal efficiency.
Our experimental results indicate that users must use the TCP mode to communicate with the server. In media sessions, a larger audio streaming or video streaming packet size is superior.
These improvements led to more than 90% successful DP2P connections. The overall user capacity in the proxy server improved 31-fold. The media relay server can serve 3524 phone calls simultaneously. The analytical results and improved methods on this study can enhance the overall performance of VoIP systems and reduce costs.


中文摘要 i
Abstract iii
致謝 v
Contents vi
List of Tables viii
List of Figures ix
Chapter 1 INTRODUCTION 1
1.1 Research Background 1
1.2 Motivation 1
1.3 Research Methods 2
1.4 Chapter Summary 2
Chapter 2 SIP Protocol 3
2.1 Introduction of the SIP Protocol 3
2.2 Instance of SIP 8
2.2.1 Registration 8
2.2.2 Call setup 10
Chapter 3 SIP Protocol Development 17
3.1 Introduction of SIP Protocol Development 17
3.2 Register Session 18
3.3 Call Setup Session 25
3.4 BYE Processes 33
Chapter 4 The NAT introduction and principle of operation 36
4.1Introduction of NAT 36
4.2 NAT Types 37
4.2.1 Full Cone 37
4.2.2 Restricted Cone 39
4.2.3 Port Restricted Cone 40
4.2.4 Symmetric Cone 41
4.3 Port Prediction Process 42
4.3.1 Introduction Prediction process 42
4.3.2 The flow-chart of Prediction process 43
4.3.3 Pre-Media Session 45
4.4 Experimental Results 48
Chapter 5 Performance Enhancement of SIP-based VoIP Server 55
5.1 Introduction 55
5.2 Evaluation on VoIP Server 55
5.3 Improvement on VoIP Server 58
5.4 Experimental Results 61
Chapter 6 Conclusion and Future 62
6.1 Conclusion 62
6.2 Future 64
Bibliography 65
Publication List 67


[1] M. Handley, and V. Jacobson, ”SDP:Session Description Protocol”, RFC2327, IETF, 1998.
[2] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, and E. Scholler, ”SIP:Session Initiation Protocol”,RFC3261,IETF,June. 2002.
[3] J. Rosenberg and H. Schulzrinne, ”An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing”, RFC3581, IETF, August 2003.
[4] Berners-Lee, ”Hypertext Transfer Protocol” ,RFC2616 ,IETF, June 1999.
[5] K. Egevang and P. Francis,”The IP Network Address Translator”, RFC1631, IETF, May 1994.
[6] P. Srisuresh and M. Holdrege,”IP Network Address Translator (NAT) Terminology and Considerations”, RFC2663, IETF, August 1999.
[7] Huitema, “Short Term NAT Requirements for UDP Based Peer-to-Peer Applications”, IETF Draft, Feb. 2001.
[8] Rosenberg, Weinberger, Huitema, Mahy, “STUN- Simple Traversal of UDP Through NATs”, RFC-3489, IETF Draft.
[9] Baruch Sterman, and David Schwartz, “NAT Traversal in SIP”, Technique Report, Delta-Three Corp., The IP Communication Network.
[10] Shiuh-Pyng Shieh, Fu-Shen Ho, Yu-Lun Hung, and Jia-Ning Luo, “Network Address Translators: Effects on Security Protocols and Applications in the TCP/IP Stack”, IEEE Internet Computing, 2000, pp.42-49.
[11] Kimchi Gur, “Traversing Firewalls and NATs”, International Application Published Under the Patent Corporation Treaty, International Patent Number: WO 02/071717 A2, 12, 2002.
[12] Khan, Md Shahadatullah, Marwood David Everett, Piche Christopher, and Chung Michael, “Method and Apparatus to Permit Data Transmission to Traverse Firewalls”, International Application Published Under the Patent Corporation Treaty, International Patent Number: WO 02/067531 A1, 2002.
[13] Wainhouse Research, LLC, Ridgeway Systems and Software, Inc, “Traversing Firewalls and NATs with Voice and Video Over IP”, 2002.
[14] Peer-to-Peer Working Group Draft, “Bi-directional Peer-to-Peer Communication with Interposing Firewalls and NATs”, IETF, 2001, pp. 1-37.
[15] Y. Takeda, “Symmetric NAT Traversal using STUN”, Internet Draft, IETF, 2003, pp. 1-23.
[16] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy, “STUN – Simple Traversal of User Datagram Protocol(UDP) Through Network Address Translators(NATs)” , RFC3489, Network Working Group, IETF, 2003, pp. 1-47.
[17] Whai-En Chen, Ya-Lin Huang, and Han-Chieh Chao, “NAT Traversing Solutions for SIP Applications”, EURASIP Journal on Wireless Communications and Networking, 2008.
[18] ITU-T, “.323: Packet-based Multimedia Communications System”, H.323, Dec., 2009.
[19] Technical Specification Group Services and System Aspects (2006), “IP Multimedia Subsystem (IMS)”, Stage 2, TS 23.228, 3rd Generation Partnership Project.
[20] S. Baset and H. Schulzrinne. “An analysis of the skype peer-to-peer Internet telephony protocol”, Columbia University Technical Report CUCS-039-04, 2004.
[21] Rosenberg, J., Weinberger, J., Huitema, C., and Mahy, R., “STUN- Simple Traversal of User Datagram Protocol (UDP)”, RFC-3489, IETF Draft, 2003.
[22] Hwang S. H. and Chung Y. H., “Modified NAT Firewall Traversal Method for SIP Communication”, US. Patent No.7751387, 2010.
[23] Takeda, Y., “Symmetric NAT Traversal using STUN”, Internet Draft, IETF, 2003, pp. 1-23.
[24] http://aws.amazon.com/ec2/
[25] Rosenberg, J., Weinberger, J., Huitema, C., and Mahy, R. ‘STUN- Simple Traversal of User Datagram Protocol (UDP)’, RFC-3489, IETF Draft, 2003.


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